Telecommunications using IP networks and VoIP technology have become a practical substitute for the traditional Public Switched Telephone Network (PSTN) that uses dedicated circuits or channels to convey telecommunications such as voice calls, facsimiles or video conferences. Unlike the PSTN model, the IP transmission model does not dedicate a network circuit or channel for a telecommunication session because a VoIP telecommunication session is transmitted via IP data packets. Two VoIP endpoints communicate directly across the IP network peer to peer. IP data packets, carrying the VoIP communication, are sent asynchronously from the source to the destination across the mesh of the IP network.
One advantage of VoIP peer to peer communications is the elimination of costly network circuits and switches dedicated to voice conversations. With VoIP technology, voice conversations can be transmitted across the same IP data network used to transmit data applications such as e-mail and web browsing. The use of a single data network for all communications and the elimination of a circuit switched networks dedicated for voice communications can offer significant savings.
A problem facing peer to peer VoIP communications, however, is the problem of global number discovery and interconnection. For example, the customers of a local telephone service provider expect to be able to send to and receive calls from any telephone number in the world. However, the telephone network of a single service provider is limited. Therefore, to provide global calling services, the local service provider must be interconnected with all other telephone networks in the world. Interconnect agreements between telephone service providers typically require a mutual billing settlement agreement for interconnect traffic exchange to ensure the terminating network is compensated for completing telephone calls.
In the traditional PTSN model, this problem was solved by routing phone calls to the switch of a dominant inter-exchange carrier (IXCs) with interconnection to any telephone number in the world. For example, the AT&T network is interconnected with the gateway switches of all national telephone provides and can ensure that a PSTN call can be routed to any telephone on Earth.
The market model for VoIP is similar to the PTSN inter-exchange model, but it is not as simple in structure. In VoIP networks, traditional telephone numbers are correlated to IP addresses. VoIP service providers must be able to route telephone calls—dialed to traditional telephone numbers—to the IP address of the VoIP provider serving the called party. A solution to this routing discovery problem has been the development of VoIP inter-exchange carriers or clearinghouses. The VoIP inter-exchange carrier provides value by facilitating the exchange of calls between hundreds of VoIP service providers. To replicate the service of the VoIP inter-exchange carrier, each VoIP service provider would have to establish a bilateral interconnect agreement with every other VoIP service provider—a huge administrative task.
FIG. 1a illustrates one solution of the conventional art. The term “conventional art” as used in this specification is not statutory “prior art” under 35 U.S.C. §102(b). This conventional art is being presented only to explain Applicant's invention in terms of technology that is existing at the filing date of this specification. Therefore, FIGS. 1a-2b do not qualify as statutory prior art.
The VoIP Carrier or Clearinghouse 100 of FIG. 1a has interconnect and settlement billing agreements with a large number of VoIP operators around the world. The VoIP Carrier or Clearinghouse operates 100 the central database which correlates the telephone numbers and IP addresses for each VoIP service provider.
While FIG. 1a shows only one source and destination VoIP network, this example applies for a very large number of VoIP service providers. When the customer of the Source VoIP Network 110 dials a phone number that cannot be completed in the Source VoIP Network 110, the Source VoIP Network 110 will send a route/access query 130 to the VoIP Carrier 100. The VoIP Carrier 100, which has the central routing database of all correlating telephone numbers and IP addresses, performs a route lookup to determine which IP address, or addresses, can terminate the VoIP call to the called number.
In the example illustrated in FIG. 1a, the route/access query 130 from the VoIP Carrier 100 returns the IP address of the Destination VoIP Network 120. Given this information, the Source VoIP Network 110 performs a call setup 140 to the destination network 120 to complete the call to the called telephone number.
It is important to note that the VoIP inter-exchange carrier or clearinghouse model differs from the circuit switched inter-exchange carrier model. In the circuit switched inter-exchange carrier model, the call is routed between the source and destination network via the inter-exchange carrier's switch. The inter-exchange switch is used for routing, inter-carrier access control and call detail record (CDR) collection for settlement billing.
In the VoIP inter-exchange carrier or clearinghouse model, the call is transmitted peer to peer from the source to the destination directly across the mesh of the IP network. There is no switch in the call path to enforce inter-carrier access control or to collect call detail records. In the VoIP model, the inter-exchange carrier or clearinghouse enforces access control by including an access token with the IP address of a destination network. The source network then includes this access token in the peer to peer call setup request to the destination network. While the destination network probably does not recognize the source network, the destination will validate the access token to determine that the call was authorized by its trusted inter-exchange carrier or clearinghouse who will guarantee payment for terminating the call.
Accounting for inter-carrier settlement billing is accomplished by call detail record collection from both the source and destination networks 110, 120. At the completion of the call, both the source and destination networks send call detail records to the VoIP inter-exchange carrier or clearinghouse. This is illustrated in FIG. 1b which shows VoIP Carrier 100 receiving a source call detail record 150a from the Source VoIP Network 110 and destination call detail record 150b from the Destination VoIP Network 120.
The previous paragraphs describe the basic peer to peer VoIP clearing and settlement call scenario. However, technology limitations and market conditions have created new variations of this basic call scenario that introduce an intermediate proxy device between the source and destination network.
FIG. 2a illustrates the presence of a VoIP Proxy Device or intermediate network 200 for peer to peer calls between a source and destination VoIP network. Common reasons for routing peer to peer VoIP calls via a proxy device or intermediate network 200 are: (A) the proxy device acts as a firewall to allow signaling of VoIP traffic between a private IP network with private IP addresses and VoIP networks with public IP addresses; (B) the source and destination networks 110, 120 use different signaling protocols and an inter-working device is required for protocol translation between the networks.
The call scenario for a VoIP inter-exchange carrier or clearinghouse 100 when a proxy or intermediate network 200 is used is illustrated in FIG. 2a and described below. When the customer of the Source VoIP Network 110 dials a phone number that cannot be completed in the Source VoIP Network 110, the Source VoIP Network 110 will send a route/access query 130a to the VoIP Carrier 100. The VoIP Carrier 100 performs a route lookup to determine which IP address, or addresses, can terminate the VoIP call to the called number.
In the example illustrated in FIG. 2a, the route/access query 130a from the VoIP Carrier 100 returns the IP address of the VoIP Proxy Device 200. Given this information, the Source VoIP Network 110 performs a call setup 140a to the VoIP Proxy Device 200. The VoIP Proxy Device 200 validates the access token in the call setup and accepts the call. If the VoIP Proxy Device cannot complete the call, it will send a route/access query 130b to the VoIP Carrier 100. The VoIP Carrier 100 performs a route lookup and determines that the route to the called number from the VoIP Proxy Device 200 can be completed by Destination VoIP Network 120.
The VoIP Carrier 100 returns the destination IP address in the route/access query response 130b to the VoIP Proxy Device 200. The VoIP Proxy Device 200 then sends a call setup 140b to the Destination VoIP Network 120 which validates the access token and completes the call.
FIG. 2b illustrates how call detail records are reported to the VoIP Carrier 100 when the call is finished. The Source VoIP Network 110 reports a source call detail record 150a for the first call leg. The VoIP Proxy Device 200 reports a destination call detail record 150c for the first call leg. The VoIP Proxy Device 200 reports a source call detail record 150d for the second call leg. The Destination VoIP Network 120 reports a destination call detail record 150b for the second call leg.
Accordingly, there is a need in the art for a more simple routing method that does not require intermediate networks to communicate with a VoIP clearinghouse to place a call. There is further need in the art for a simplified routing method that can reduce a number of call detail records reported to a VoIP clearinghouse from one or more intermediate networks used to place a call.